The present invention is related to the field of media processing, such as processing of voice or other audio signals in a telephone network.
Media processing is employed when it is desirable to perform some kind of transformation on a digital representation of an analog signal. In telephone or other voice systems, for example, it is common to apply a compression transformation on digital voice signals in order to improve transmission efficiency by reducing bandwidth requirements. A corresponding de-compression transformation is applied to received compressed signals. Other common examples include various forms of filtering and “scrambling”, or encoding for purposes of security.
Devices through which multiple channels of voice or other analog signals flow typically employ one or more digital signal processors (DSPs) to carry out the desired processing. Hardware and/or software within the device is responsible for assigning channels to DSPs, steering received data of the various channels to the appropriate DSP(s) for processing, and re-combining the processed channel data in some form for re-transmission to another device. The number of DSPs employed at a given device is determined in part by the number of channels and the expected processing load per channel. If the processing algorithm for one channel consumes all the processing capacity of a DSP, then it is generally necessary to have one DSP per channel. If each DSP has sufficient processing capacity to handle the processing load for multiple channels, then fewer DSPs are required.
In conventional telephone systems, which are based on time-division-multiplexing (TDM) technology, the processing of multiple channels by a single DSP takes a special form. In TDM systems, all of the channels are synchronous with respect to each other, and therefore it is a simple matter to divide the use of a DSP into a given number of time slots and allocate the time slots to respective channels. There is a certain rigidity in the operation, however, that tends to favor a particular manner of use of the DSP. Whatever frame interval is employed for collecting sufficient channel data for a quantum of processing, there is no benefit from a latency perspective to processing the data frame in any time less than the frame interval, because there is no opportunity to output the processed frame until an entire frame interval has passed. Given that the latency through a TDM device is determined by the signalling format rather than the actual processing time, and that the relative timing of the channels is so well known, it makes sense to simply load each DSP with the maximum number of channels it can handle while meeting real-time constraints, i.e., while processing frames at least as fast as frames are provided to the DSP. This approach makes full utilization of each DSP, promoting efficiency as measured in a “cost per channel” sense.
In recent times, techniques have evolved for transmitting voice and other media data over non-synchronous networks such as Internet Protocol (IP) networks. In contrast to TDM networks, there is no necessary timing relationship among different “channels”, or distinct streams of media data. Even when the nominal data rates of different channels are the same, which would be the case for example for channels carrying voice encoded according to the same encoding algorithm, there is considerable variability in the relative arrival times of packets of the different channels at a given network device. At one instant, it may be that packets of channels 1, 2 and 3 are received sequentially in that order, while at another instant, they may all be received substantially simultaneously or even in reverse order.
It may be desirable in certain applications that a network node add only a minimal amount of latency to media streams. This may be the case, for example, if a node is operating in series with other equipment that adds considerable latency that approaches an overall end-to-end latency goal for a channel. In such cases, it would be desirable for packets to be processed as quickly as possible, without wasting time in buffers for purposes of synchronization with a DSP or other packet streams. However, the prevailing architecture for processing multiple media channels forces each stream to be processed by a fully-loaded DSP, which can result in latency far above what might be desired in a given system. It would be desirable to process packet media streams in a manner that would enable stricter latency goals to be met.